Re: Asterisk on Freerunner was: voip on Debian
On Wednesday 29 April 2009, Nicola Mfb wrote: > 2009/4/19 Nicola Mfb : > > 2009/4/19 Al Johnson : > > [...] > > As AMI emits all needed events I'll add fso support for the GUI to > > handle the switching automatically, while for a true voip fso > > [...] > > I added fso support to switch between stereoout when ringing and > voip-handset when the call is established but asterisk does not reacts > well on this and stop to capture audio. > It works well if I set the voip scenario before launching it and never > switches to stereoout. > Before digging again in the asterisk alsa code I'd like to know if the > scenario switching is transparent to alsa applications, or may brings > underrun/overrun or other problems that needs to be managed in a > stronger way. Scenario switching ought to be transparent to apps, but that might not be true if there's a change in the 'DAI mode' setting. There's more on this in the wiki: http://wiki.openmoko.org/wiki/Neo_1973_audio_subsystem I don't have the state files too hand to see if this is being changed, but it's the only setting I can think of that might upset an app. Can you reload chan_alsa after the state change? I don't remember how granular the asterisk reload options are, but it might be a quick'n'dirty workaround. ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: Asterisk on Freerunner was: voip on Debian
2009/4/19 Nicola Mfb : > 2009/4/19 Al Johnson : > [...] > As AMI emits all needed events I'll add fso support for the GUI to > handle the switching automatically, while for a true voip fso [...] I added fso support to switch between stereoout when ringing and voip-handset when the call is established but asterisk does not reacts well on this and stop to capture audio. It works well if I set the voip scenario before launching it and never switches to stereoout. Before digging again in the asterisk alsa code I'd like to know if the scenario switching is transparent to alsa applications, or may brings underrun/overrun or other problems that needs to be managed in a stronger way. Nicola ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: Asterisk on Freerunner was: voip on Debian
2009/4/26 Rask Ingemann Lambertsen : > On Sat, Apr 18, 2009 at 05:49:05PM +0200, Nicola Mfb wrote: > >> I will be happy to write an AMI gui but now I'm hold having problems >> with the alsa channel. Using the pcm default is not compatible with >> the default shipped /etc/asound.conf, so I just tried to use >> plughw:dnsoop and plughw:dmix, the result is that there freerunner >> does not ring on incoming call (and you cannot hear the other peer), >> while audio transmitting is perfect. Using plughw:0,0 for input/output >> works but I have stuttered audio (from freerunner to peer). > > Why are you not using hw:0,0? Asterisk has fixed-hardcoded settings for alsa (8000hz, 1 channel etc), and they are incompatible using hw directly, plughw autoconvert sound streams but it uses very short buffer/period size so the stuttered audio (I guess). Nicola ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: Asterisk on Freerunner was: voip on Debian
On Sat, Apr 18, 2009 at 05:49:05PM +0200, Nicola Mfb wrote: > I will be happy to write an AMI gui but now I'm hold having problems > with the alsa channel. Using the pcm default is not compatible with > the default shipped /etc/asound.conf, so I just tried to use > plughw:dnsoop and plughw:dmix, the result is that there freerunner > does not ring on incoming call (and you cannot hear the other peer), > while audio transmitting is perfect. Using plughw:0,0 for input/output > works but I have stuttered audio (from freerunner to peer). Why are you not using hw:0,0? -- Rask Ingemann Lambertsen Danish law requires addresses in e-mail to be logged and stored for a year ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: Asterisk on Freerunner was: voip on Debian
2009/4/24 Timo Juhani Lindfors : > Nicola Mfb writes: >> But I'm happy, asterisk runs fine in a real case. > > Can you check if you get lower latency by only running linphone on fr > and having the 3g stick connected to fr itself? I cannot before next tuesday, but during the weekend I'll test FR connected to ADSL router directly with wifi. I'm quite sure that playing with asound.conf will fix the high latency, as using asterisk with direct plughw:0,0 (short period/buffer size) gived stuttered alsa capture, but near realtime output playback Nicola ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: Asterisk on Freerunner was: voip on Debian
Nicola Mfb writes: > But I'm happy, asterisk runs fine in a real case. Can you check if you get lower latency by only running linphone on fr and having the 3g stick connected to fr itself? ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: Asterisk on Freerunner was: voip on Debian
> (I'm just thinking how many om guys got the same in the last two years! :) > LOL, just as many as distros and alsa states here :) Great work > ___ > Openmoko community mailing list > community@lists.openmoko.org > http://lists.openmoko.org/mailman/listinfo/community > -- David Reyes Samblas Martinez http://www.tuxbrain.com Open ultraportable & embedded solutions Openmoko, Openpandora, GP2X the Wiz, Letux 400, Arduino Hey, watch out!!! There's a linux in your pocket!!! ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: Asterisk on Freerunner was: voip on Debian
2009/4/21 Nicola Mfb : > 2009/4/19 Nicola Mfb : [...] > I'll update about my progress on AMI interface soon. It's great night for me! I was able to do my first VoIP->PSTN call with FR, it was to my girlfriend of course, It may be for love or It may be to not bother some other guy with an unpredictible test :) I used for that my all damned pre-pre-pre-alpha tools I'm writing (and hope to finish). The test case is interesting, please be quite with comments, I'm crazy, not mad :) I'm from Paduli a small village where I spend my weekends, there I have ADSL with a voip option to call flat Italy landlines, during the week I'm far in Naples for my job, there I have only an umts card. To use voip I have to be connected phisically to the ADSL router, no use is permitted from public internet, and my provider uses a modifyed sip protocol. And now the test scenario. In Paduli: *) atheros openwrt/kamikaze powered embedded device up 24h *) it's connected to a stupid adsl router I cannot change/reflash as Telecom Italia uses the non standard sip protocol with a secret virtual channel for voip. *) openvpn server with tap layer 2 to make external connections appear as in LAN :) In Naples: *) laptop connected to internet with E220 HSDPA *) freerunner connected to laptop acting as router with BT/Bnep (testing my bt manager) *) freerunner connected to Paduli LAN with openvpn client *) runned alice-ctl, a tool to fake a Telecom cordless able to connect to the voip service, based on pivelli python code (I rewrote it in C before as python did not fit in my embedded atheros device!) *) alice-ctl enabled a peer on my vpn IP (acting as the fake cordless) *) asterisk acted as the cordless, built with two patch, the first to speak the tampered SIP protocol (thanks again to pivelli project), the second to solve the announced alsa problems *) launched my very very rude voip dialer that interacts with asterisk trough the AMI interface and finally placed the Call! And now the results: The call was picked up from my girlfriend father, the result was: "Hello... Emh are you there... Yes umh. do you hear me?... Yes but it's strange" -> "Papi give me the phone!, it was Nicola with freerunner for sure" :))) (I'm just thinking how many om guys got the same in the last two years! :) A big delay, but superb audio quality, we stayed up for about 15 mins. I think that it's only a problem due of my absourd networking and asound.conf tuning as the period/buffer size is huge for a good latency... But I'm happy, asterisk runs fine in a real case. There is a lot to do, please join and contribute, I will happy tho share everything! Regards Nicola ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: Asterisk on Freerunner was: voip on Debian
thanks a lot :D 2009/4/22 Nicola Mfb > 2009/4/21 kimaidou : > > Hi > > thanks for this feedback ! > > Could you please write a wiki page about this, if not already done ? > > I started a page at http://wiki.openmoko.org/wiki/Asterisk > Everyone interested is invited to correct (english is not my native > language) and collaborate, there is a lot to do :) > > Regards > > Nicola > > ___ > Openmoko community mailing list > community@lists.openmoko.org > http://lists.openmoko.org/mailman/listinfo/community > ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: Asterisk on Freerunner was: voip on Debian
2009/4/21 kimaidou : > Hi > thanks for this feedback ! > Could you please write a wiki page about this, if not already done ? I started a page at http://wiki.openmoko.org/wiki/Asterisk Everyone interested is invited to correct (english is not my native language) and collaborate, there is a lot to do :) Regards Nicola ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: Asterisk on Freerunner was: voip on Debian
Hi thanks for this feedback ! Could you please write a wiki page about this, if not already done ? Thanks again Kimaidou 2009/4/21 Nicola Mfb > 2009/4/19 Nicola Mfb : > > Some alsa guru may take a look at the chan_alsa.c file of asterisk > 1.4.17? > > Jaroslav Kysela of ALSA pointed me to the problem (thanks), and > effectively asterisk code does not support dmix plugin in it's state, > I corrected it with a fast 2 line change workaround working only with > dmix (a real patch is needed), and now asterisk/alsa is working great > and stable. > It's about two hours that I'm listening some mp3 on it sent by ekiga > on the laptop :) > I'll update about my progress on AMI interface soon. > >Nicola > > ___ > Openmoko community mailing list > community@lists.openmoko.org > http://lists.openmoko.org/mailman/listinfo/community > ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: Asterisk on Freerunner was: voip on Debian
2009/4/19 Nicola Mfb : > Some alsa guru may take a look at the chan_alsa.c file of asterisk 1.4.17? Jaroslav Kysela of ALSA pointed me to the problem (thanks), and effectively asterisk code does not support dmix plugin in it's state, I corrected it with a fast 2 line change workaround working only with dmix (a real patch is needed), and now asterisk/alsa is working great and stable. It's about two hours that I'm listening some mp3 on it sent by ekiga on the laptop :) I'll update about my progress on AMI interface soon. Nicola ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: Asterisk on Freerunner was: voip on Debian
2009/4/20 Esben Stien : > Nicola Mfb writes: > >> But we may superseed on this actually until having a well working >> asterisk on freerunner > > Rather definitely use freeswitch;). Hi Esben, Actually only a patch for asterisk let me use the voip line provided by my adsl carrier (Alice/Telecom Italia) as it uses a modified sip protocol. For that reason I did not take a look at freeswitch. However I'm curious to know if someone used freeswitch on freerunner, is there some OE recipe to build it? Nicola ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: Asterisk on Freerunner was: voip on Debian
Nicola Mfb writes: > But we may superseed on this actually until having a well working > asterisk on freerunner Rather definitely use freeswitch;). -- Esben Stien is b...@e s a http://www. s tn m irc://irc. b - i . e/%23contact sip:b0ef@ e e jid:b0ef@n n ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: Asterisk on Freerunner was: voip on Debian
2009/4/19 Nicola Mfb :
[...]
> Some alsa guru may take a look at the chan_alsa.c file of asterisk 1.4.17?
Here a little c snippet to show you easily the problem (that I have on
the desktop too). So it seems an alsa-lib bug/feature ?
#include
int main(int argc, char **argv) {
int err;
snd_pcm_t *handle;
struct pollfd pfd;
fd_set fds;
err = snd_pcm_open(&handle, "default" ,
SND_PCM_STREAM_PLAYBACK, O_NONBLOCK);
if (err < 0) {
puts("snd_pcm_open error");
exit(1);
}
err = snd_pcm_poll_descriptors_count(handle);
if (err != 1) {
puts("snd_pcm_poll_descriptors_count problem");
exit(1);
}
snd_pcm_poll_descriptors(handle, &pfd, err);
FD_ZERO(&fds);
FD_SET(pfd.fd,&fds);
err=select(pfd.fd+1,NULL,&fds,NULL,NULL);
if (err<1) puts("select failed");
puts("Ok");
}
Save the above line in alsatest.c and compile with cc -o alsatest
alsatest.c -lasound,
launch it and if "default" is dmixed you well not see "Ok".
Changing default in plughw:0,0 or hw:0,0 in alsatest.c recompile and
you'll see the "Ok" immediately e.g. the behaviour that asterisk would
like!
Regards
Nicola
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Re: Asterisk on Freerunner was: voip on Debian
2009/4/19 Nicola Mfb : > 2009/4/19 Al Johnson : > [...] > I have stuttered outgoing audio, so I think the problem is with alsa > buffer/periods etc., the proposed asound.conf file should work as > create longer buffer/periods both for input and output, but using it > asterisk does not speak anymore to the erapiece. I enabled the logging > to maximum details and I may see effectively "Alsa/default is ringing" > but no sound is emitted, and I do not know why, as in this situation > aplay works fine. I did some test and think (hope) isolated the problem. Asterisk get an fd poll descriptor from alsa playback pcm, when a sound has to be emitted this fd is added to the write fdset. The alsa thread loops around select() and when the alsa driver is ready to receive more frames to write the select is waken and asterisk effectively send out the sound. Now that's working only if using plughw:0,0 as the output device, with some sort of dmix, multiplexer etc, the select will never be waken by the write fd descriptor so asterisk will never emit any sound. Some alsa guru may take a look at the chan_alsa.c file of asterisk 1.4.17? Nicola ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: Asterisk on Freerunner was: voip on Debian
On Sunday 19 April 2009, Nicola Mfb wrote: > 2009/4/19 Al Johnson : > > For linphone I use Brian Code's asound.conf : > >http://www.koolu.org/asound.conf > > This uses dmix and dsnoop and gives stutter-free sound in both directions > > with linphone. It does have echo since we can't use the suppression in > > the GSM chipset. > > Asterisk should be able to do echo suppression? Potentially, assuming it doesn't use too much CPU. The linphone echo suppression option didn't seem to do anything, but I've not looked at why. > Is this present with external headphones too? It shouldn't be, but I've not tried it. It might have buzz if GSM is transmitting too. ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: Asterisk on Freerunner was: voip on Debian
2009/4/19 Al Johnson : [...] >> Let's survive this interesting topic. >> I will be happy to write an AMI gui but now I'm hold having problems >> with the alsa channel. Using the pcm default is not compatible with >> the default shipped /etc/asound.conf, so I just tried to use >> plughw:dnsoop and plughw:dmix, the result is that there freerunner >> does not ring on incoming call (and you cannot hear the other peer), >> while audio transmitting is perfect. > > I'm guessing 'does not ring' means it uses the earpiece for ringing instead of > the speaker. You will need stereoout.state for the ringing, then change to > voip-handset.state when answering the call. This is what is needed when > working with linphone, although the change of state is not automated yet. > voip-handset.state is in both FSO and SHR IIRC. You should be able to do the > state switch with an asterisk script. As AMI emits all needed events I'll add fso support for the GUI to handle the switching automatically, while for a true voip fso integration I think that's has to be discussed as actually on incoming GSM call fso will automatically switch to stereoout and gsm alsa state and may create problems during a voip session. But we may superseed on this actually until having a well working asterisk on freerunner, and I'll complete the AMI gui. However if I use plughw:0,0 in asterisk alsa.conf I may hear the ring in the earpiece (the only problem was that I had to change the speaker alsa control from 0 to max in voip state). I have stuttered outgoing audio, so I think the problem is with alsa buffer/periods etc., the proposed asound.conf file should work as create longer buffer/periods both for input and output, but using it asterisk does not speak anymore to the erapiece. I enabled the logging to maximum details and I may see effectively "Alsa/default is ringing" but no sound is emitted, and I do not know why, as in this situation aplay works fine. [...] > For linphone I use Brian Code's asound.conf : > http://www.koolu.org/asound.conf > This uses dmix and dsnoop and gives stutter-free sound in both directions with > linphone. It does have echo since we can't use the suppression in the GSM > chipset. Asterisk should be able to do echo suppression? Is this present with external headphones too? regards Nicola ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: Asterisk on Freerunner was: voip on Debian
On Saturday 18 April 2009, Nicola Mfb wrote: > 2008/9/6 TL Mieszkowski : > > I've had a lot of success running both twinkle and asterisk and I thought > > I'd share my experiences. > > Twinkle works well, but the gui is limiting on the touchscreen. I think > > once configured properly > > asterisk will make an excellent voip backend for the neo. You can > > control it through asterisk > > manager commands by writing text strings to a socket, and which has hooks > > for most languages I'm sure. > > Let's survive this interesting topic. > I will be happy to write an AMI gui but now I'm hold having problems > with the alsa channel. Using the pcm default is not compatible with > the default shipped /etc/asound.conf, so I just tried to use > plughw:dnsoop and plughw:dmix, the result is that there freerunner > does not ring on incoming call (and you cannot hear the other peer), > while audio transmitting is perfect. I'm guessing 'does not ring' means it uses the earpiece for ringing instead of the speaker. You will need stereoout.state for the ringing, then change to voip-handset.state when answering the call. This is what is needed when working with linphone, although the change of state is not automated yet. voip-handset.state is in both FSO and SHR IIRC. You should be able to do the state switch with an asterisk script. > Using plughw:0,0 for input/output > works but I have stuttered audio (from freerunner to peer). I tried > the mentioned asound.conf from koolu too, the same, If i move out from > plughw there is no sound in fr with asterisk. If I use dnsoop form > input and plughw for output, the input is stuttered again. I'm using > shr-testing and asterisk 1.4.17-r1 from the same branch. > > As in the old thread there was success story may someone share some hint? For linphone I use Brian Code's asound.conf : http://www.koolu.org/asound.conf This uses dmix and dsnoop and gives stutter-free sound in both directions with linphone. It does have echo since we can't use the suppression in the GSM chipset. ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Asterisk on Freerunner was: voip on Debian
2008/9/6 TL Mieszkowski : > > I've had a lot of success running both twinkle and asterisk and I thought I'd > share my experiences. > Twinkle works well, but the gui is limiting on the touchscreen. I think > once configured properly > asterisk will make an excellent voip backend for the neo. You can control > it through asterisk > manager commands by writing text strings to a socket, and which has hooks > for most languages I'm sure. Let's survive this interesting topic. I will be happy to write an AMI gui but now I'm hold having problems with the alsa channel. Using the pcm default is not compatible with the default shipped /etc/asound.conf, so I just tried to use plughw:dnsoop and plughw:dmix, the result is that there freerunner does not ring on incoming call (and you cannot hear the other peer), while audio transmitting is perfect. Using plughw:0,0 for input/output works but I have stuttered audio (from freerunner to peer). I tried the mentioned asound.conf from koolu too, the same, If i move out from plughw there is no sound in fr with asterisk. If I use dnsoop form input and plughw for output, the input is stuttered again. I'm using shr-testing and asterisk 1.4.17-r1 from the same branch. As in the old thread there was success story may someone share some hint? Regards Nicola ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: voip on Debian
Alastair Johnson ha scritto: > Marco Trevisan (Treviño) wrote: > Was it you who mentioned having patched linphone to switch alsa states , > and to tweak the GUI to fit the screen better? Yes I was, but my work isn't complete yet :P Unfortunately I've to write my bits in too many places... :| -- Treviño's World - Life and Linux http://www.3v1n0.net/ ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: voip on Debian
Marco Trevisan (Treviño) wrote:
> Alastair Johnson ha scritto:
>> Marco Trevisan (Treviño) wrote:
>>> Alastair Johnson wrote:
Marco Trevisan (Treviño) wrote:
> TL Mieszkowski wrote:
>> alsactl -f /usr/share/openmoko/scenarios/voip-handset.state restore
> About this, using this alsa control file, can you get the caller voice
> only in the earpiece?
> If I use it in a Om2008 I get the voice both in the earpiece and in the
> main speaker!
>
This will do as you say since control.3 is for the headset/speaker, and
control.4 for the handset earpiece. I don't know if this is the result
of someone committing an old file or what.
>>> Ok, the problem is that if I invert the values I heard voice in the
>>> earpiece, but I continue hearing it also in the main speaker!
>>> It's really annoying! :|
>> I'm surprised unless it's a small amount of bleedthrough. Setting both
>> channels of control.3 to 0 should silence it. This should silence the
>> speaker entirely:
>>
>> control.94 {
>> comment.access 'read write'
>> comment.type BOOLEAN
>> comment.count 1
>> iface MIXER
>> name 'Amp Spk Switch'
>> value false
>> }
>>
>>
>>> Anyone got it working correctly? In this situation VoIP is not usable... :(
>> See:
>> http://wiki.openmoko.org/wiki/Linphone
>> This points to http://www.koolu.org/voip-handset.state which I have used
>> successfully with the CLI version of linphone compiled using
>> mokomakefile a while back. FDOM currently has a more recent linphone
>> with GUI and presumably a working voip-handset.state too.
>
> Thanks. Now it finally work. I had already tried that state file from
> koolu but I wasn't able to make it work! :o
> After the edit you suggested me the incoming audio works well. I'm using
> linphone 2.1.1 with a gui too, I compiled it long time ago, but I always
> had this kind of problem!
Was it you who mentioned having patched linphone to switch alsa states ,
and to tweak the GUI to fit the screen better?
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Re: voip on Debian
Alastair Johnson ha scritto:
> Marco Trevisan (Treviño) wrote:
>> Alastair Johnson wrote:
>>> Marco Trevisan (Treviño) wrote:
TL Mieszkowski wrote:
> alsactl -f /usr/share/openmoko/scenarios/voip-handset.state restore
About this, using this alsa control file, can you get the caller voice
only in the earpiece?
If I use it in a Om2008 I get the voice both in the earpiece and in the
main speaker!
>>> This will do as you say since control.3 is for the headset/speaker, and
>>> control.4 for the handset earpiece. I don't know if this is the result
>>> of someone committing an old file or what.
>> Ok, the problem is that if I invert the values I heard voice in the
>> earpiece, but I continue hearing it also in the main speaker!
>> It's really annoying! :|
>
> I'm surprised unless it's a small amount of bleedthrough. Setting both
> channels of control.3 to 0 should silence it. This should silence the
> speaker entirely:
>
> control.94 {
> comment.access 'read write'
> comment.type BOOLEAN
> comment.count 1
> iface MIXER
> name 'Amp Spk Switch'
> value false
> }
>
>
>> Anyone got it working correctly? In this situation VoIP is not usable... :(
>
> See:
> http://wiki.openmoko.org/wiki/Linphone
> This points to http://www.koolu.org/voip-handset.state which I have used
> successfully with the CLI version of linphone compiled using
> mokomakefile a while back. FDOM currently has a more recent linphone
> with GUI and presumably a working voip-handset.state too.
Thanks. Now it finally work. I had already tried that state file from
koolu but I wasn't able to make it work! :o
After the edit you suggested me the incoming audio works well. I'm using
linphone 2.1.1 with a gui too, I compiled it long time ago, but I always
had this kind of problem!
--
Treviño's World - Life and Linux
http://www.3v1n0.net/
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Re: voip on Debian
Marco Trevisan (Treviño) wrote:
> Alastair Johnson wrote:
>> Marco Trevisan (Treviño) wrote:
>>> TL Mieszkowski wrote:
alsactl -f /usr/share/openmoko/scenarios/voip-handset.state restore
>>> About this, using this alsa control file, can you get the caller voice
>>> only in the earpiece?
>>> If I use it in a Om2008 I get the voice both in the earpiece and in the
>>> main speaker!
>>>
>> I just checked the file and it is clearly wrong as it says:
>>
>> control.3 {
>> comment.access 'read write'
>> comment.type INTEGER
>> comment.count 2
>> comment.range '0 - 127'
>> iface MIXER
>> name 'Headphone Playback Volume'
>> value.0 127
>> value.1 127
>> }
>> control.4 {
>> comment.access 'read write'
>> comment.type INTEGER
>> comment.count 2
>> comment.range '0 - 127'
>> iface MIXER
>> name 'Speaker Playback Volume'
>> value.0 0
>> value.1 0
>> }
>>
>> This will do as you say since control.3 is for the headset/speaker, and
>> control.4 for the handset earpiece. I don't know if this is the result
>> of someone committing an old file or what.
>
> Ok, the problem is that if I invert the values I heard voice in the
> earpiece, but I continue hearing it also in the main speaker!
> It's really annoying! :|
I'm surprised unless it's a small amount of bleedthrough. Setting both
channels of control.3 to 0 should silence it. This should silence the
speaker entirely:
control.94 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 1
iface MIXER
name 'Amp Spk Switch'
value false
}
> Anyone got it working correctly? In this situation VoIP is not usable... :(
See:
http://wiki.openmoko.org/wiki/Linphone
This points to http://www.koolu.org/voip-handset.state which I have used
successfully with the CLI version of linphone compiled using
mokomakefile a while back. FDOM currently has a more recent linphone
with GUI and presumably a working voip-handset.state too.
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Re: voip on Debian
Alastair Johnson wrote:
> Marco Trevisan (Treviño) wrote:
>> TL Mieszkowski wrote:
>>> alsactl -f /usr/share/openmoko/scenarios/voip-handset.state restore
>> About this, using this alsa control file, can you get the caller voice
>> only in the earpiece?
>> If I use it in a Om2008 I get the voice both in the earpiece and in the
>> main speaker!
>>
>
> I just checked the file and it is clearly wrong as it says:
>
> control.3 {
> comment.access 'read write'
> comment.type INTEGER
> comment.count 2
> comment.range '0 - 127'
> iface MIXER
> name 'Headphone Playback Volume'
> value.0 127
> value.1 127
> }
> control.4 {
> comment.access 'read write'
> comment.type INTEGER
> comment.count 2
> comment.range '0 - 127'
> iface MIXER
> name 'Speaker Playback Volume'
> value.0 0
> value.1 0
> }
>
> This will do as you say since control.3 is for the headset/speaker, and
> control.4 for the handset earpiece. I don't know if this is the result
> of someone committing an old file or what.
Ok, the problem is that if I invert the values I heard voice in the
earpiece, but I continue hearing it also in the main speaker!
It's really annoying! :|
Anyone got it working correctly? In this situation VoIP is not usable... :(
--
Treviño's World - Life and Linux
http://www.3v1n0.net/
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Re: voip on Debian
Marco Trevisan (Treviño) wrote:
> TL Mieszkowski wrote:
>> 1.) You need the alsa state for voip handset. Can be got here:
>>
>> http://svn.openmoko.org/trunk//src/target/audio/om-gta02/
>>
>> This goes in /usr/share/openmoko/scenarios/
>> load it with the command :
>>
>> alsactl -f /usr/share/openmoko/scenarios/voip-handset.state restore
>
> About this, using this alsa control file, can you get the caller voice
> only in the earpiece?
> If I use it in a Om2008 I get the voice both in the earpiece and in the
> main speaker!
>
I just checked the file and it is clearly wrong as it says:
control.3 {
comment.access 'read write'
comment.type INTEGER
comment.count 2
comment.range '0 - 127'
iface MIXER
name 'Headphone Playback Volume'
value.0 127
value.1 127
}
control.4 {
comment.access 'read write'
comment.type INTEGER
comment.count 2
comment.range '0 - 127'
iface MIXER
name 'Speaker Playback Volume'
value.0 0
value.1 0
}
This will do as you say since control.3 is for the headset/speaker, and
control.4 for the handset earpiece. I don't know if this is the result
of someone committing an old file or what.
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Re: voip on Debian
TL Mieszkowski wrote: > 1.) You need the alsa state for voip handset. Can be got here: > > http://svn.openmoko.org/trunk//src/target/audio/om-gta02/ > > This goes in /usr/share/openmoko/scenarios/ > load it with the command : > > alsactl -f /usr/share/openmoko/scenarios/voip-handset.state restore About this, using this alsa control file, can you get the caller voice only in the earpiece? If I use it in a Om2008 I get the voice both in the earpiece and in the main speaker! -- Treviño's World - Life and Linux http://www.3v1n0.net/ ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: voip on Debian
TL Mieszkowski wrote: > On Thu, Oct 2, 2008 at 8:44 AM, Davide Scaini > <[EMAIL PROTECTED]> wrote: >> Ekiga? did you tryed that on [EMAIL PROTECTED] >> so curious! > > I haven't, but I see no reason why it wouldn't work. It doesn't do IAX > though, only SIP. And sip is problematic behind a NAT firewall. Use an outbound proxy like siproxd and the problems go away. ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: voip on Debian
On Thu, Oct 2, 2008 at 8:44 AM, Davide Scaini <[EMAIL PROTECTED]> wrote: > Ekiga? did you tryed that on [EMAIL PROTECTED] > so curious! I haven't, but I see no reason why it wouldn't work. It doesn't do IAX though, only SIP. And sip is problematic behind a NAT firewall. > On Thu, Oct 2, 2008 at 2:56 PM, Esben Stien <[EMAIL PROTECTED]> wrote: >> >> Asterisk is dead. Long live freeswitch. >> >> Didn't you get the memo?;) >> I was under the impression that freeswitch was more geared to low level operations, carrier level stuff.(?) -- View this message in context: http://n2.nabble.com/voip-on-Debian-tp842903p1134642.html Sent from the Openmoko Community mailing list archive at Nabble.com. ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: voip on Debian
Ekiga? did you tryed that on [EMAIL PROTECTED] so curious! d On Thu, Oct 2, 2008 at 2:56 PM, Esben Stien <[EMAIL PROTECTED]> wrote: > TL Mieszkowski <[EMAIL PROTECTED]> writes: > > > asterisk will make an excellent voip backend for the neo > > Asterisk is dead. Long live freeswitch. > > Didn't you get the memo?;) > > -- > Esben Stien is [EMAIL PROTECTED] s a > http://www. s tn m > irc://irc. b - i . e/%23contact > sip:b0ef@ e e > jid:b0ef@n n > > ___ > Openmoko community mailing list > community@lists.openmoko.org > http://lists.openmoko.org/mailman/listinfo/community > ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: voip on Debian
TL Mieszkowski <[EMAIL PROTECTED]> writes: > asterisk will make an excellent voip backend for the neo Asterisk is dead. Long live freeswitch. Didn't you get the memo?;) -- Esben Stien is [EMAIL PROTECTED] s a http://www. s tn m irc://irc. b - i . e/%23contact sip:b0ef@ e e jid:b0ef@n n ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: voip on Debian
As of now, I'm using asterisk on debian to connect to an IAX2 provider. (diamoncard.us) There is a far end echo, that is being caused by asterisk on the Freerunner. Other than that, it is working perfectly. I don't know much about the FSO framework or zhone, but it would be trivial (from the asterisk end) to use zhone as the front end, as it would only take sending asterisk manager commands to control the console. Configuration of asterisk with a gui would be a sticking point, but not too complicated by any means. I'm still working on eliminating the echo, but I have a feeling that nothing short of a recompile will work (or a channel driver for the wolfson codec instead of using alsa, both of which I am ignorant about). I saw the asterisk .ipk in the community repo, does anyone know the source of that package, or who compiled it? -- View this message in context: http://n2.nabble.com/voip-on-Debian-tp842903p1132559.html Sent from the Openmoko Community mailing list archive at Nabble.com. ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: voip on Debian
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Interesting. Good find and setup. I am currently working on some iax2 related code for gta02 and future phones as far as the voip stack. I was running into some audio problems for awhile, but am starting to get all those little bugs worked out. There was ideas of a really slimmed down version of Asterisk being the _actual_ client in which people will use. This will allow them to use some of the core functionality of Asterisk as well. We will see where it goes. I am pretty familiar with Asterisk Channel drivers :) And it all depends on how easily the technology you are hooking into is. - -bk Florian Hackenberger wrote: | On Saturday 06 September 2008, TL Mieszkowski wrote: |> There is the potential to do some really cool stuff with asterisk it |> has quite a bit of functionality. | | We should really write a channel driver for the Neo (wolfson codec & GSM | modem daemon). We could then use asterisk for custom voicemail boxes, | dialplan routing (think time based blacklists etc.) for calls coming in | over GSM. As far as I'm familiar with the asterisk channel modules, | that should not be too difficult. | | Cheers, | Florian | -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFI4uQLWSn2Kv7ZyAoRAqMXAJ9GhiSR7jvObO+vYDlMC/Kx5b0FpQCghUNt 0C25mer+0w2/uPS7OULUhNs= =H8zz -END PGP SIGNATURE- ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: voip on Debian
On Saturday 06 September 2008, TL Mieszkowski wrote: > There is the potential to do some really cool stuff with asterisk it > has quite a bit of functionality. We should really write a channel driver for the Neo (wolfson codec & GSM modem daemon). We could then use asterisk for custom voicemail boxes, dialplan routing (think time based blacklists etc.) for calls coming in over GSM. As far as I'm familiar with the asterisk channel modules, that should not be too difficult. Cheers, Florian -- DI Florian Hackenberger [EMAIL PROTECTED] www.hackenberger.at ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
voip on Debian
I've had a lot of success running both twinkle and asterisk and I thought I'd share my experiences. Twinkle works well, but the gui is limiting on the touchscreen. I think once configured properly asterisk will make an excellent voip backend for the neo. You can control it through asterisk manager commands by writing text strings to a socket, and which has hooks for most languages I'm sure. The difficult part is getting a good set of configuration files for asterisk. I think for the most part I have a good setup for sip. iax could be configured too (important I think for the encryption). Heres the steps as well as I can remember: 1.) You need the alsa state for voip handset. Can be got here: http://svn.openmoko.org/trunk//src/target/audio/om-gta02/ This goes in /usr/share/openmoko/scenarios/ load it with the command : alsactl -f /usr/share/openmoko/scenarios/voip-handset.state restore 2.) Install asterisk, or twinkle, (or whatever, I got those two to work). In any program other than asterisk, you must enter your sip server info. For asterisk you need these changes: -modules.conf: change the sound module from oss to alsa (about halfway down) -alsa.conf: uncomment the audio devices and use `plughw` as the devices instead of `hw` like this "input_device=plughw:0,0" set autoanswer=no -sip.conf: you need to set your realm for your sip server set an outbound sip registration: register => user:[EMAIL PROTECTED] set authentication credentials for outgoing calls: auth=user:[EMAIL PROTECTED] I recommend using disallow=all & allow=ulaw or alaw, to avoid stressing the cpu, unless you have a slow net connection. -extensions.conf: you need to set up extensions to forward to your sip service exten => _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED]) It really helps to have an asterisk server that isn't NATed to test with If someone out there has the skills to make a gui, I can do the backend asterisk stuff. There is the potential to do some really cool stuff with asterisk it has quite a bit of functionality. -- View this message in context: http://n2.nabble.com/voip-on-Debian-tp842903p842903.html Sent from the Openmoko Community mailing list archive at Nabble.com. ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
